asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	2364626811	res_pjsip_t38: Don't pass T.38 control frames through to other hooks. This crept up during gateway testing where the gateway would receive the request to negotiate and assume it came from the remote side, causing the gateway state machine to go a little, to a use a technical term, "wonky". ........ Merged revisions 403364 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	d61f258384	Initialize the hash value argument to pj_hash_get() to 0. Passing a non-zero value causes PJLIB to use the given input as the hash value. Passing zero causes the parameter to become an output parameter that receives the hash value that was computed based on the given key. This change essentially makes ast_sip_dict_get() properly retrieve the desired value. ........ Merged revisions 403349 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403350 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b8025e789d	res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag. Newer versions of PJSIP have changed to using a flag for the PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a configure check to detect the presence of the flag and use it if found. ........ Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	3357c494cb	sorcery, bucket: Change observer remove calls to take const callbacks struct. * Make ast_sorcery_observer_remove() accept a const callbacks struct. * Make ast_sorcery_observer_remove() tolerant of the sorcery parameter being NULL. Now it can be called within a module unload routine if the sorcery initialization fails. * Fix ast_sorcery_observer_add() to fail if the container link fails. ........ Merged revisions 403324 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403327 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	177e7861a2	res_pjsip_session: Apply fromuser and fromdomain to all requests as documented. ........ Merged revisions 403271 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403272 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	88c840db50	res_pjsip_t38: Add the framehook to the channel only on first INVITE. The check for determining whether the T.38 framehook should be added to the channel or not has now been changed to guarantee adding only occurs on the first incoming or outgoing INVITE. ........ Merged revisions 403258 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	0620cc0c00	res_pjsip_transport_websocket: Fix security events and simplify implementation. Transport type determination for security events has been simplified to use the type present on the message itself instead of searching through configured transports to find the transport used. The actual WebSocket transport has also been simplified. It now leverages the existing PJSIP transport manager for finding the active WebSocket transport for outgoing messages. This removes the need for res_pjsip_transport_websocket to store a mapping itself. (closes issue ASTERISK-22897) Reported by: Max E. Reyes Vera J. Review: https://reviewboard.asterisk.org/r/3036/ ........ Merged revisions 403256 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	e93fbf41e6	res_ari: Add Recording events to the validator. ........ Merged revisions 403240 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	56290895aa	res_pjsip_sdp_rtp: Don't produce an invalid media stream with no formats. Depending on configuration it was possible for a media stream to be created without any media formats. The produced SDP would fail internal validation and cause a crash. The code will now no longer add media streams with no formats to the SDP, allowing it to pass validation and work. (closes issue ASTERISK-22858) Reported by: Anthony Messina ........ Merged revisions 403223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	b315b16c90	res_pjsip_header_funcs: Don't add headers to re-INVITEs. When sending a re-INVITE to an endpoint it was possible for received headers to be added as well (since they are stored for retrieval using the PJSIP_HEADER dialplan function). This caused a broken (and potentially large) SIP INVITE to be produced and sent. This changes the module so it will no longer add headers to re-INVITEs. (closes issue ASTERISK-22882) Reported by: David M. Lee ........ Merged revisions 403221 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403222 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	6019353ad6	res_stasis_playback: Add 'number', 'digits', and 'characters' URI scheme implementations. This change adds new URI scheme implementations for playing numbers, digits, and characters. This is done as part of the normal playback mechanism and can be used with queueing to create a combined sentence. Review: https://reviewboard.asterisk.org/r/3028/ ........ Merged revisions 403209 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403210 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	a64cd7c6bb	res_pjsip_session: Add configurable behavior for redirects. The action taken when a redirect occurs is now configurable on a per-endpoint basis. The redirect can either be treated as a redirect to a local extension, to a URI that is dialed through the Asterisk core, or to a URI that is dialed within PJSIP itself. (closes issue ASTERISK-21710) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2963/ ........ Merged revisions 403207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403208 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	9d3b590ad8	res_pjsip: Fix crash when reloading certain configurations. Certain options available that specify a SIP URI perform validation on the provided URI using the PJSIP URI parser. This operation requires that the thread executing it be registered with the PJLIB library. During reloads this was done on a thread which was NOT registered with it. This fixes the problem by creating a task which reloads the configuration on a PJSIP thread. (closes issue ASTERISK-22923) Reported by: Anthony Messina ........ Merged revisions 403179 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403180 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	fccb427c88	ari:Add application/json parameter support The patch allows ARI to parse request parameters from an incoming JSON request body, instead of requiring the request to come in as query parameters (which is just weird for POST and DELETE) or form parameters (which is okay, but a bit asymmetric given that all of our responses are JSON). For any operation that does _not_ have a parameter defined of type body (i.e. "paramType": "body" in the API declaration), if a request provides a request body with a Content type of "application/json", the provided JSON document is parsed and searched for parameters. The expected fields in the provided JSON document should match the query parameters defined for the operation. If the parameter has 'allowMultiple' set, then the field in the JSON document may optionally be an array of values. (closes issue ASTERISK-22685) Review: https://reviewboard.asterisk.org/r/2994/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	fd33969240	res_pjsip: Update handling of some options to work with new option names. Some options (such as call_group and pickup_group) share the same configuration handler and decide what logic to use based on the name of the option. These handlers were not updated to check for the new option names and were treating the options as invalid. This change simply updates the handlers with the proper names of the options. (closes issue ASTERISK-22922) Reported by: Anthony Messina ........ Merged revisions 403173 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403174 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	ed48377994	ARI: Implement device state API Created a data model and implemented functionality for an ARI device state resource. The following operations have been added that allow a user to manipulate an ARI controlled device: Create/Change the state of an ARI controlled device PUT /deviceStates/{deviceName}&{deviceState} Retrieve all ARI controlled devices GET /deviceStates Retrieve the current state of a device GET /deviceStates/{deviceName} Destroy a device-state controlled by ARI DELETE /deviceStates/{deviceName} The ARI controlled device must begin with 'Stasis:'. An example controlled device name would be Stasis:Example. A 'DeviceStateChanged' event has also been added so that an application can subscribe and receive device change events. Any device state, ARI controlled or not, can be subscribed to. While adding the event, the underlying subscription control mechanism was refactored so that all current and future resource subscriptions would be the same. Each event resource must now register itself in order to be able to properly handle [un]subscribes. (issue ASTERISK-22838) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3025/ ........ Merged revisions 403134 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	05cbf8df9b	res_pjsip: AMI commands and events. Created the following AMI commands and corresponding events for res_pjsip: PJSIPShowEndpoints - Provides a listing of all pjsip endpoints and a few select attributes on each. Events: EndpointList - for each endpoint a few attributes. EndpointlistComplete - after all endpoints have been listed. PJSIPShowEndpoint - Provides a detail list of attributes for a specified endpoint. Events: EndpointDetail - attributes on an endpoint. AorDetail - raised for each AOR on an endpoint. AuthDetail - raised for each associated inbound and outbound auth TransportDetail - transport attributes. IdentifyDetail - attributes for the identify object associated with the endpoint. EndpointDetailComplete - last event raised after all detail events. PJSIPShowRegistrationsInbound - Provides a detail listing of all inbound registrations. Events: InboundRegistrationDetail - inbound registration attributes for each registration. InboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowRegistrationsOutbound - Provides a detail listing of all outbound registrations. Events: OutboundRegistrationDetail - outbound registration attributes for each registration. OutboundRegistrationDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsInbound - A detail listing of all inbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. PJSIPShowSubscriptionsOutbound - A detail listing of all outboundbound subscriptions and their attributes. Events: SubscriptionDetail - on each subscription detailed attributes SubscriptionDetailComplete - raised after all detail records have been listed. (issue ASTERISK-22609) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2959/ ........ Merged revisions 403131 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	14a7452934	ari: Add events for playback and recording. While there were events defined for playback and recording these were not actually sent. This change implements the to_json handlers which produces them. (closes issue ASTERISK-22710) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/3026/ ........ Merged revisions 403119 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	eda7126862	ari: Add Snoop operation for spying/whispering on channels. The Snoop operation can be invoked on a channel to spy or whisper on it. It returns a channel that any channel operations can then be invoked on (such as record to do monitoring). (closes issue ASTERISK-22780) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3003/ ........ Merged revisions 403117 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	76a2b855e1	res_pjsip: convert configuration settings names to snake case some more Updated the alembic script for pjsip. Also, the dtls config parsing stuff was expecting strings with no underscores, so removed the underscores from the option name before passing it to the parser. ........ Merged revisions 403082 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403083 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	d9015a5356	ARI: Don't leak implementation details This change prevents channels used as implementation details from leaking out to ARI. It does this by preventing creation of JSON blobs of channel snapshots created from those channels and sanitizing JSON blobs of bridge snapshots as they are created. This introduces a framework for excluding information from output targeted at Stasis applications on a consumer-by-consumer basis using channel sanitization callbacks which could be extended to bridges or endpoints if necessary. This prevents unhelpful error messages from being generated by ast_json_pack. This also corrects a bug where BridgeCreated events would not be created. (closes issue ASTERISK-22744) Review: https://reviewboard.asterisk.org/r/2987/ Reported by: David M. Lee ........ Merged revisions 403069 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1c45a32ee8	res_pjsip: convert configuration settings names to snake case Renamed, where appropriate, the configuration options for chan/res_pjsip to use snake case (compound words separated by an underscore). For example, faxdetect will become fax_detect, recordofffeature will become record_off_feature, etc... Review: https://reviewboard.asterisk.org/r/3002/ ........ Merged revisions 403022 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	92af2b2e26	res_pjsip_session: Fix memory leak of direct media format capabilities The direct media format capabilities are always allocated in ast_sip_session_alloc and were not freed in the session destructor. Whoops. (This being the third whoops caught by Scott and Nitesh's valgrind work for the Asterisk Test Suite. Nifty!) ........ Merged revisions 402968 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402969 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c3575e338e	res_pjsip_sdp_rtp: Fix use of uninitialized value in PJSIP In PJMEDIA, pjmedia_sdp_rtpmap_to_attr will attempt to use the string rtpmap.param regardless of its length value. Simply setting the length to 0 does not prevent the garbage on the stack in rtpmap.param.ptr from being formatted in a sprintf call. This patch initializes the string to NULL so that at the very least, something is provided to the function that is predictable. ........ Merged revisions 402941 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402943 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	08686e62c5	res_pjsip_mwi: Fix memory leak of MWI subscriptions container This patch fixes a reference counting memory leak on the ao2_container created as part of create_mwi_subscriptions. When we create the container in this routine, the intent is to hand lifetime ownership over to the global container unsolicited_mwi. When ao2_global_obj_replace_unref is called, the reference count on mwi_subscriptions (the container) will be bumped by 1; however, the function does not decrement the reference count on mwi_subscriptions when this occurs. This will prevent the container from being fully disposed of when Asterisk exits (or on any subsequent call to this operation, such as during a reload). ........ Merged revisions 402940 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402942 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	f0ccc59a22	stasis: Fixed scoping problem with bridge tracking. ........ Merged revisions 402817 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402929 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	d1ad4a95f8	ari: Add silence generator controls This patch adds the ability to start a silence generator on a channel via ARI. This generator will play silence on the channel (avoiding audio timeouts on the peer) until it is stopped, or some other media operation is started (like playing media, starting music on hold, etc.). (closes issue ASTERISK-22514) Review: https://reviewboard.asterisk.org/r/3019/ ........ Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	71612fb007	res_pjsip_caller_id: Don't overwrite user portion of the From header when fromuser is set. The fromuser option is used to explicitly set the user within the From header. The res_pjsip_caller_id module did not take this setting into account when determining if the From header could be modified or not. (closes issue ASTERISK-22866) Reported by: Anthony Messina ........ Merged revisions 402891 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402892 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	1b14a78d14	res_pjsip: Add support for building against pjproject with SIP transaction group lock support. SIP transaction group lock support has been backported into our pjproject. Since the code now internally uses a group lock the code is now changed to unlock it if present. Note that the act of finding the transaction is what actually returns it locked. For further information about group locks check out the wiki page at: http://trac.pjsip.org/repos/wiki/Group_Lock (issue ASTERISK-22818) Reported by: Matt Jordan ........ Merged revisions 402864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	67b650543c	res_ari_channels: Add the ability to stop locally generated ringing on a channel. Using the 'ring' operation it is possible to start locally generated ringback if the channel is answered. This change adds the ability to stop it by using DELETE. ........ Merged revisions 402804 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	f6593b4156	ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404 Was returning a 404 on a valid technology with an empty list of endpoints. Now checking against the channel tech to make sure the tech itself is valid and not just an empty list of endpoints. (issue ASTERISK-22803) Reported by: David M. Lee ........ Merged revisions 402793 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4be01b4598	ari endpoints: GET /ari/endpoints/{invalid-tech} should return a 404 Implementation listing endpoints by technology returned an empty array if no matching endpoints were found. Fixed so a "404 Not Found" will be returned instead. (closes issue ASTERISK-22803) Reported by: David M. Lee ........ Merged revisions 402787 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	12a0edac69	pjsip_messaging, pjsip_header_funcs: Crashes due to NULL pointer dereferences Both res_pjsip_messaging and res_pjsip_header_funcs were causing asterisk to crash because they were trying to dereference a NULL pointer. In the case of res_pjsip_messaging it was attempting to "print" a contact header that did not exist. In fact contact headers should not be part of a SIP MESSAGE, so the offending code was simply removed. In the case of res_pjsip_header_funcs a null private channel tech was being passed to the function and then later dereferenced. Added null checks (and error logging) to the read/write function handlers to guard against crashing. (closes issue ASTERISK-22821) Reported by: Anthony Messina ........ Merged revisions 402757 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402758 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	5701c51bb6	Fixed a typ. ........ Merged revisions 402738 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	7323809607	res_stasis.c: Fix locking issues with the app_bridge_moh container. * Fix unlinking from the app_bridges_moh container in remove_bridge_moh() without a lock under normal circumstances. * Made check ast_bridge_set_after_callback() return value in bridge_moh_create() to handle failure. * Fixed SCOPED_AO2LOCK() locking over too much scope in stasis_app_bridge_moh_channel() and stasis_app_bridge_moh_stop(). * Fixed unusual usage of ao2_unlink_flag() in control_unlink(). * Fixed orphaned bridge from off nominal path in stasis_app_bridge_create(). * Fixed strange construct in stasis_app_unsubscribe(). From a bad merge? * Made load_module() cleanup on failure. Review: https://reviewboard.asterisk.org/r/2962/ ........ Merged revisions 402593 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	518f091a1a	Clarify an ambiguous error message. ........ Merged revisions 402582 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	4c128198c8	res_pjsip: Print a helpful error message if sorcery registration fails ........ Merged revisions 402570 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402572 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	b83a3965b8	Changes from make ari-stubs after r402560 ........ Merged revisions 402561 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402571 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4f1bdeed1c	ARI playback: Rename ARI Playback to Playbacks Before playback was the only non plural resource. It has been renamed to playbacks for consistency. (closes issue ASTERISK-22737) Reported by: Paul Belanger ........ Merged revisions 402560 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	97a8debd90	ari: Add application/x-www-form-urlencoded parameter support ARI POST calls only accept parameters via the URL's query string. While this works, it's atypical for HTTP API's in general, and specifically frowned upon with RESTful API's. This patch adds parsing for application/x-www-form-urlencoded request bodies if they are sent in with the request. Any variables parsed this way are prepended to the variable list supplied by the query string. (closes issue ASTERISK-22743) Review: https://reviewboard.asterisk.org/r/2986/ ........ Merged revisions 402555 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402557 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	3c645e8520	PJSIP: Improve error handling in digest authenticator Previously, regardless of whether failure to authenticate was due to lacking any authentication or actually failing authentication, the Digest Authenticator would simply return that a challenge was still needed. It will continue to do that when no authentication information is in the received SIP digest, but when authentication information is present and does not pass authentication, that will be treated as an authentication error. This is to ensure that PJSIP will issue security events indicated failed auths. ........ Merged revisions 402537 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7d0d1a1efb	ari: User better nicknames for ARI operations While working on building client libraries from the Swagger API, I noticed a problem with the nicknames. channel.deleteChannel() channel.answerChannel() channel.muteChannel() Etc. We put the object name in the nickname (since we were generating C code), but it makes OO generators redundant. This patch makes the nicknames more OO friendly. This resulted in a lot of name changing within the res_ari_.so modules, but not much else. There were a couple of other fixed I made in the process. When reversible operations (POST /hold, POST /unhold) were made more RESTful (POST /hold, DELETE /unhold), the path for the second operation was left in the API declaration. This worked, but really the two operations should have been on the same API. * The POST /unmute operation had still not been REST-ified. Review: https://reviewboard.asterisk.org/r/2940/ ........ Merged revisions 402528 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7678fd040e	res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media. The ring operation sends ringing to the specified channel it is invoked on. The dtmf operation can be used to send DTMF digits to the specified channel of a specific length with a wait time in between. Finally hangup reasons allow you to specify why a channel is being hung up (busy, congestion). Early media behavior has also been tweaked slightly. When playing media to a channel it will no longer automatically answer. If it has not been answered a progress indication is sent instead. (closes issue ASTERISK-22701) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2916/ ........ Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	98dea21bc1	chan_sip: Fix RTCP port for SRFLX ICE candidates This corrects one-way audio between Asterisk and Chrome/jssip as a result of Asterisk inserting the incorrect RTCP port into RTCP SRFLX ICE candidates. This also exposes an ICE component enumeration to extract further details from candidates. (closes issue ASTERISK-21383) Reported by: Shaun Clark Review: https://reviewboard.asterisk.org/r/2967/ ........ Merged revisions 402345 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 402348 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402349 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	4053f36a71	res_ari_channels: Fix a deadlock when originating multiple channels close to eachother. If a Stasis application is specified an implicit subscription is done on the originated channel. This was previously done with the channel lock held which is dangerous as the underlying code locks the container and iterates items. This change releases the lock on the originated channel before subscribing occurs. (closes issue ASTERISK-22768) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2979/ ........ Merged revisions 402346 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d17a780333	res_stasis: Ensure the channel is always departed from the bridge when it leaves. This change adds a command to the command queue to explicitly depart the channel from the bridge when it is told it has left. If the channel has already been departed or has entered a different bridge this command will become a no-op. (closes issue ASTERISK-22703) Reported by: John Bigelow (closes issue ASTERISK-22634) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2965/ ........ Merged revisions 402336 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	069da1e75a	stasis: add functions embarrassingly missing from r400522 I neglected to implement two of the endpoint subscription functions when I did the work. Normally, you'll only hit that when you unsubscribe from a specific endpoint. ........ Merged revisions 402276 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402277 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	4746c068dc	pjsip_messaging: Added debug for in dialog messaging (issue ASTERISK-22777) Reported by: Matt Jordan ........ Merged revisions 402265 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	aa7f9e55f2	ARI: Remove channels/{channelId}/dial This removes the /ari/channels/{channelId}/dial URI since it is redundant, overly complex, is likely to become more externally complex over time, and is too high-level compared with other ARI operations. See the following for further information: http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000002.html (closes issue ASTERISK-22784) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2968/ ........ Merged revisions 402152 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402153 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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2314 Commits (236462681192cfde8bd3e17b3a3e4ebbcbcb2c13)